[asterisk-scf-commits] asterisk-scf/release/media_rtp_pjmedia.git branch "master" updated.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Tue Sep 27 10:20:58 CDT 2011
branch "master" has been updated
via ca14499ac6e6bf66f3eb5a33bfc4d1a12f6452d2 (commit)
from 79856c7897e1d9552c43e7d368b7c7f0268e3f3d (commit)
Summary of changes:
config/Rtp.config | 56 +++++++++++++++++++++++++++++++++++++++++++++++++++++
1 files changed, 56 insertions(+), 0 deletions(-)
- Log -----------------------------------------------------------------
commit ca14499ac6e6bf66f3eb5a33bfc4d1a12f6452d2
Author: Brent Eagles <beagles at digium.com>
Date: Tue Sep 27 12:50:15 2011 -0230
Comments for the ICE and SRTP related configuration options.
diff --git a/config/Rtp.config b/config/Rtp.config
index e1a99dd..47b975d 100644
--- a/config/Rtp.config
+++ b/config/Rtp.config
@@ -16,3 +16,59 @@ workerthreadcount=4
# IPv6 address we should bind sessions to
#ipv6bind=
+
+#
+# Enable SRTP support for RTP. If not enabled, attempts to allocate
+# an SRTP media session will fail. Enabled by default.
+#
+# enablesrtp=false
+
+#
+# Configure ICE support for NAT.
+#
+[rtpoverice]
+
+#
+# Enable the ICE transport. Disbled by default.
+#
+# rtpovericeenable=false
+
+#
+# Enable including TURN as a NAT traversal option. Default false.
+#
+# rtpovericewithturn=false
+
+#
+# Set the limits for the number of candidates that will be sussed out. It is
+# usually okay to leave this unset.
+#
+# rtpicemaxcandidates=10
+
+#
+# Set the maximum number of ICE sessions supported. An ICE media flow
+# is essentially a negotiated and managed flow between two endpoints, so
+# there is some resource consumption per connection.
+#
+# rtpicemaxcalls=50
+
+#
+# Configure the STUN server hostname. Note that some public STUN servers may
+# not reply in a timely manner so you may need to experiment here.
+#
+# stunserverhost=[insert STUN server host here]
+
+#
+# Configure the STUN server port number. You can usually leave this blank as
+# the default would suffice.
+#
+# stunserverport=3478
+
+#
+# Configure the TURN server hostname.
+#
+# turnserverhost=[insert TURN server host here]
+
+#
+# Configure the TURN server port
+#
+# turnserverport=3478
-----------------------------------------------------------------------
--
asterisk-scf/release/media_rtp_pjmedia.git
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